search for: u1000

...matrix(c(1000,5000,5000,5000,2150, 0,1000,5000,5000,1750, 0,0,1000,5000,1150, 0,0,0,1000,200, 1000,1000,5000,5000,2050, 0,1000,1000,5000,1972), ncol=5, byrow=T) colnames(a70)=c(paste("x", 1:4, sep=""), "med") a70 <- as.data.frame(a70) start=800; end=2000 step=10; u1000=start-step u1000 <- u1000+step # varying the 1000 entry a70[a70==1000] <- u1000 reg70 <- lm(a70$med ~ -1+x1+x2+x3+x4, data=a70) res <- sum( (reg70$residuals^2) ) for (i in 1:( (end-start)/step) ){ a70[a70==u1000] <- u1000+step u1000 <- u1000+step reg70 <...

Hi Lee, You need to specify the VM context that you are using.. so using your examples.. extensions.conf entry.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000) exten => 1000,102,Voicemail2(b1000) exten => 1000,103,Hangup should be.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000@sip) exten => 1000,102,Voicemail2(b1000@sip) exten => 1000,103,Hangup and.. exten => 8500,1,Wait,1 exten => 8500,2,VoiceMailMain2(...

...; want asterisk in the media stream for features type=friend ; haven't tried this by creating a user and peer for each handset (yuck) extensions.conf entries are either one of the following (tested against both) exten => _XXXX,1,Dial(SIP/1000,30,t) exten => _XXXX,2,Voicemail(u1000) exten => _XXXX,3,Hangup exten => _XXXX,102,Voicemail(b1000) exten => _XXXX,103,Hangup exten => _XXXX,1,Dial(SIP/1000,30,t) exten => _XXXX,2,Voicemail(u1000) exten => _XXXX,3,Hangup exten => _XXXX,102,Dial(SIP/1001,30,t) exten => _XXXX,103,Voicemail(u1000) exten => _XXXX...

...e command string passed to bash to call Rscript and package.skeleton() followed by a cat of the NAMESPACE file. ----------------------------------------------------------------------------- edd at rob:/tmp$ mkdir skeleton edd at rob:/tmp$ cd skeleton/ edd at rob:/tmp/skeleton$ docker run --rm -ti -u1000:1000 -v ${PWD}:/mnt -w /mnt r-base:3.6.3 bash -c "Rscript -e 'x <-1L; package.skeleton(name=\"p363\")'; cat p363/NAMESPACE" Creating directories ... Creating DESCRIPTION ... Creating NAMESPACE ... Creating Read-and-delete-me ... Saving functions and data ... Making he...

...audio available on SIP/66.147.170.34-0811abe8?? Here is my exten map [actual phone number munged]. I have removed the Grandstream from the exten for this example. It makes no difference whether the Grandstream gets rang or not: exten => 9725551212,1,Answer exten => 9725551212,2,Voicemail2(u1000) exten => 9725551212,3,Hangup Also, just for testing, I have added this extension: exten => 2501,1,Voicemail2(u1000) exten => 2501,2,Hangup If I dial 2501 from my grandstream, voicemail works that way, too. My questions: 1) Should I have the Answer in there or not? It doesn't hel...

...the same time I allow the phone to ring before trying another phone in the queue. Is there a way to tell asterisk....? If this call is coming from a queue, do not follow a normal dial plan for the phone (don't send to user's voicemail). In stead, once timed out (t|||60), send to Voicemail(u1000). Lee recommended QUEUESTATUS, but that seems to return if anyone is in a specific queue, and not if the current call came from a queue. I probably just misunderstand how it all works. :) Thanks all! Rob ----------------- I would recommend that you download the following tool and play with it...

....,3,Dial(SIP/rix/${EXTEN}|20|t) exten => _0.,4,Congestion exten => _0.,104,Congestion [sip-in] exten => 1000,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP}) exten => 1000,2,Monitor(wav,${CALLFILENAME},m) exten => 1000,3,Dial(SIP/alex,20) exten => 1000,4,Voicemail(u1000) -- Alex Polite http://polite.se

Here's what I'm looking to do.... exten => 10,1,Dial(SIP/1000&SIP/1001,15,wW) exten => 10,2,Voicemail(u1000) This should ring both phones and they should keep ringing for the alloted time before moving on. However, it appears that if one of the phones is Busy, it returns with a busy and moves on without really ringing the second phone. Short of checking if the channels are available or using a queue,...

...h to call Rscript and > package.skeleton() followed by a cat of the NAMESPACE file. > > ----------------------------------------------------------------------------- > edd at rob:/tmp$ mkdir skeleton > edd at rob:/tmp$ cd skeleton/ > edd at rob:/tmp/skeleton$ docker run --rm -ti -u1000:1000 -v ${PWD}:/mnt -w /mnt r-base:3.6.3 bash -c "Rscript -e 'x <-1L; package.skeleton(name=\"p363\")'; cat p363/NAMESPACE" > Creating directories ... > Creating DESCRIPTION ... > Creating NAMESPACE ... > Creating Read-and-delete-me ... > Saving functio...

...=0.0 txgain=4.0 immediate=no musiconhold=default busydetect=no callprogress=no usecallerid=uk channel => 1 -------------------------------- part of extensions.conf [incoming] exten => s,1,SetCallerId(${CALLERID}) exten => s,2,dial(SIP/5000&SIP/5001,10,tr) exten => s,3,Voicemail,u1000 exten => s,102,Voicemail,b1000 exten => _9.,1,Dial(${CONSOLE}/${EXTEN:1}) exten => _9.,2,Congestion Vassilis

...e (a Cisco 7940). I've got a call-waiting feature on my line and couldnt figure out how to trigger a flash in order to go from one call to another. Solution : 1st - The inbound context (in extensions.conf of course) [pstninbound] exten=>s,1,Dial(SIP/cisco7940,40|Tt) exten=>s,2,Voicemail(u1000) exten=>s,3,Voicemail(b1000) The Tt option will allow you to transfer by hitting the # key. 2nd - The flash extension Now, somewhere in your extensions file, create a context that similar to this : exten=>604,1,Flash() exten=>604,2,Dial(SIP/cisco7940) By transfering a ZAP call to tha...

...ode=inband canreinvite=no qualify=yes user=phone [200] type=friend secret=010101 auth=md5 nat=yes host=dynamic reinvite=no canreinvite=no dtmfmode=inband callerid="Fred F"<200> dissallow=all Extensions.conf [default] exten => 1000,1,Dial,Zap/1|20 exten => 1000,2,Voicemail,u1000 exten => 1000,3,Hangup exten => 1000,102,Voicemail,b1000 exten => 1000,103,Hangup exten => 2000,1,Dial,Zap/2|20 exten => 2000,2,Voicemail,u2000 exten => 2000,3,Hangup exten => 2000,102,Voicemail,b2000 exten => 2000,103,Hangup exten => _NXXNXXXXXX, 1, dial(SIP/${EXTEN}@s...

...And then the phone rejects the call based on a malformed From: header. The from-pstn context is really simple at this point: [from-pstn] ; CALLS FROM PSTN NETWORK ; ;exten => s,1,VoicemailMain2 exten => s,1,NoOp(${CALLERID}) exten => s,2,Dial(${SIP_PHONE},15,j) exten => s,3,Voicemail(u1000) ; Gavin's voicemail, unavailable exten => s,4,Hangup exten => s,103,Voicemail(b1000) ; On the phone exten => s,104,Hangup My Zapata.conf file hasn't changed since 1.0.x and looks like the following: [channels] ; entries are inherited ; added this to see if it...

I have a question about how Asterisk Parses the Dial Plan. To create a hunt-group which would be the appropriate dial plan: [CompanyABC] exten => 7228888,1,Dial(SIP/8017228888,60,r) exten => 7228888,102,Dial(SIP/8014361234,60,r) exten => 7228888,103,Dial(SIP/8014362345,60,r) exten => 7228888,104,Dial(SIP/8014363456,60,r) exten => 7228888,105,Dial(SIP/8014364567,60,r) exten

...t; s,4,BackGround(VY-ThanksForCalling); Play VY intro message (daytime) ;exten => s,5,Goto(s,4) exten => i,1,Playback(invalid) ; "That's not valid, try again" exten => i,2,Goto(s,4) exten => t,1,Playback(vm-goodbye) exten => t,2,Hangup exten => 2,1,Voicemail(u1000) exten => 5,1,Directory(VYStaff) exten => 7,1,ResponseTimeout(1) exten => 7,2,Voicemail(u70) <-LONG PAUSE COMES HERE exten => 7,3,Playback(vm-goodbye) exten => 7,4,Hangup Asterisk CLI output: -- Set Response Timeout to 1 -- Executing VoiceMail("Zap/1-1&...

...clude => ciscophones exten => s,1,Wait,1 exten => s,2,Answer exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,15 exten => s,5,BackGround(auto-greeting) exten => s,6,BackGround(auto-menu) [ciscophones] exten => 1000,1,Dial(SIP/1000,15) exten => 1000,2,Voicemail(u1000) exten => 1001,1,Dial(SIP/1001,15) exten => 1001,2,Voicemail(u1001) exten => o,1,Goto(incoming,s,6) exten => 0,1,Goto(ciscophones,1001,1) Now if I press "0" during the voicemail prompt - it will Dial extension 1001 instead of routing to the [incoming] context and re-play th...

Dear all, I have a question regarding the configuration of *. I have 3 port FXS, and 2 port FXO. I have 4 users that use analog phone connected to FXS (I have 3 phones). I need to limit the user's capability (user A can call International, user B can call long distance, etc). I want to implement the password say to call , he/she needs to puch 9(for outgoing call)2-4 digits password,then

...{VARIABLE}} or ${text${VARIABLE}} or any hybrid ; [globals] CONSOLE=/dev/dsp [voicemail] exten => 7,1,Ringing exten => 7,2,Wait(2) exten => 7,3,VoicemailMain [local] include => voicemail ; SIP Phone Operator Office exten => 1000,1,Dial,SIP/opoffice|30 exten => 1000,2,Voicemail,u1000 exten => 1000,102,Voicemail,b1000 ; SIP Phone Operator Field exten => 2000,1,Dial,SIP/opfield|30 exten => 2000,2,Voicemail,u2000 exten => 2000,102,Voicemail,b2000 approach 2: ======== I moved the opoffice Snom to the office side and moved the configurations for this phone in /etc/s...

Hi all! Please I need someone that have already done an H.323 - SIP gateway to help me with some problems. I can stablish calls from a SIP telephone to a H.323, but I can't do vice versa... (problems with port 1719- when the gatekeeper tries to contact with asterisk at this port, it is unrecheable...). Please someone can help me? Regards, Mireia

...rivacyManager exten => 1000/_635.,2,Macro(startrecord,${EXTEN},${CALLERIDNUM}) exten => 1000,2,Wait(0) exten => 1000,3,Wait(0) ; yes i know I dont need this exten => 1000,4,Set(GROUP()=PSTN) ;Set Group exten => 1000,5,Dial(SIP/spa-1,20,TWtw) exten => 1000,6,Voicemail(u1000) exten => 1000,7,Hangup Thanks, Ryan